Re: [Asterisk-Users] SIP register problem
Karl Brose wrote: Is the SIPquest server sending the 401 Unauthorized message verbatim as you printed it here? I.e. is the WWW-Authentcate header broken up into several lines like that? If so, how man spaces are actually at the beginning of each new line? Continuation lines are allowed in SIP, but I think it's only one space at the beginning of each new line. To make Asterisk parse this correctly you need to turn on pedantic=yes It's silly that Asterisk doesn't turn this header parsing on by default, no reason not to. I agree. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium E100P or TE410P card
We are located in Belgium and just ordered a PRA line, the telco asked the following questions : - 120 or 75 ohm ? - Support for CRC4 yes/no ? - B channels in 2 way ? We will buy a digium card but which one should we buy ? could anybody help me with this ? Thank you Michael DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100 or TE410 card an PRA line
We are located in Belgium and just ordered a PRA line, the telco asked the following questions : - 120 or 75 ohm ? - Support for CRC4 yes/no ? - B channels in 2 way ? We will buy a digium card but which one should we buy ? could anybody help me with this ? Thank you Michael Sorry for the previous html mail DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interrupting MusicOnHold while call in queue ?
You can either do a make samples (which will overwrite your existing configuration) or look in the configs subdirectory in the asterisk source tree for queues.conf.sample. It would be nicer if make samples would install the configs as .sample, for example: /etc/asterisk/iax.conf.sample. That way it's safe to do make samples (which is always a good thing) and it's easy to add changes because all you have to do is add these *.sample files to CVS and do a cvs diff -u after you've installed a new version. That way new lines and changes in defaults are spotted and can be put in the live configuration files. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: AW: [Asterisk-Users] Voice in Asterisk with BRI ISDN Anyproperworking configurations yet?
Hi, I have a HFC based ISDN BRI card in a Fedora Core 2 box (2.6.5 kernel). I was just wondering, is zaphfc the best way to interface this type of card with Asterisk? I've managed to get all other types of interfacing on Asterisk going except for BRI ISDN. I'd would really like to get BRI ISDN going with a HFC card since they are very inexpensive cards. Any advice would be greatly appreciated! Kind Regards. Christiaan Brink Systems Developer Molo Afrika Speech Technologies (Pty) Ltd. (Cell) +2782 410 7370 (Tel) +2712 346 3336 (Fax) +2712 346 3337 South Africa -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin List-Petersen Sent: Friday, November 19, 2004 3:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: AW: [Asterisk-Users] Voice in Asterisk with BRI ISDN Anyproperworking configurations yet? On Thu, 2004-11-18 at 14:35, Pascal C. Kocher wrote: Hello Tim I'm struggling to get a HFC card running in NT mode. It seems to work for a short period but then stops. The messages on the asterisk console mentione something about event 6. Is the zaphfc module enough to be loaded or must hisax also be loaded in order to work? zaphfc works fine in NT mode, no problems here (Ackermann Euracom P4 and Teles.FON). Don't load the HiSax driver and zaphfc, that would not work. If you have another card (like Teles, Winbond etc.) that you need the HiSax driver for, leave the support for the hfc based cards out during the kernel-compile. You don't need it, when using zaphfc. The reason, why it stops (like no dialtone anymore) is when you run ztcfg more than once, after the machine has booted. That'll kill zaphfc and is a known bug. Slán leat, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] compiling error
Hi, I tried to compiled Asterisk on a Mandrake 9.2 and I encountered this error command: make after compiling for sometime then this error appeared gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o utils.o editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv -lssl /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make: *** [asterisk] Error 1 I'm not really knowledgeable in compiling. What does this mean? Did I missed something? TIA, Wesley ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] compiling error
On Fri, 2004-11-19 at 16:40 +0800, Wesley Jay Deypalan wrote: Hi, I tried to compiled Asterisk on a Mandrake 9.2 and I encountered this error /usr/bin/ld: cannot find -lssl I'm not really knowledgeable in compiling. What does this mean? Did I missed something? Missing lib SSL, or more likely the -dev or -devel package. Error messages are usually fairly simple, like this. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] i swtiched to digest
I believe the list is so big that many of us are loosing some interesting threads. May be the admins can split the users list in some more specific sub-lists, and the people who wants to receive all the messages can subscribe to the sublists, or have a digest for someones, etc. Regards -- Pablo Povarchik Quality Colocation and Dedicated Servers services Colocation facilities include Fremont California, London UK and Trento Italy +--- FuturaHost.Com - Industrial Business Class ISP + | Web Hosting - Dedicated Servers - Colocation | [EMAIL PROTECTED] - http://futurahost.com/ - (+39) 0461 592710 | Get a high quality full cabinet with 5Mbps full burst included |for only ¤700/month, availability also in London +-+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ericsson or ACC - AXC or Tigris ??
Hi folks, just wondering if there might be any users of these devices on the lists. particularly if you are using version 12.5 software. Gary . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] compiling error
Read Asterisk install, You need to install libssl package Wesley Jay Deypalan wrote: Hi, I tried to compiled Asterisk on a Mandrake 9.2 and I encountered this error command: make after compiling for sometime then this error appeared gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o utils.o editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv -lssl /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make: *** [asterisk] Error 1 I'm not really knowledgeable in compiling. What does this mean? Did I missed something? TIA, Wesley ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] linux distribution
Hi everybody: Please send me your recommenation of the best fit linux version for Asterisk application. Is there a one stop web-site where I can download everything. __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Config files (was: Re: [Asterisk-Users] Interrupting MusicOnHold while call in queue ?)
On Fri, Nov 19, 2004 at 07:34:19PM +1100, Edwin Groothuis wrote: You can either do a make samples (which will overwrite your existing configuration) or look in the configs subdirectory in the asterisk source tree for queues.conf.sample. It would be nicer if make samples would install the configs as .sample, for example: /etc/asterisk/iax.conf.sample. That way it's safe to do make samples (which is always a good thing) and it's easy to add changes because all you have to do is add these *.sample files to CVS and do a cvs diff -u after you've installed a new version. That way new lines and changes in defaults are spotted and can be put in the live configuration files. http://bugs.digium.com/bug_view_page.php?bug_id=0002908 -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR Question
How can I tell the dialled number from CDR records? We need to be able to bill our provider based on the dialled number. Is this possible? Ben Merrills Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Little off topic
Martin List-Petersen wrote: You can't, the T100P is a unchannelized T1 card. This is 100% wrong. The T100P supports Channelized Voice T-1 (aka CT1) If you want to use it with HylaFax you need either SpanDSP OR an analog port on Asterisk in addition to the T100P. A search of the mailing lists would have told you this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100 or TE410 card an PRA line
Michael Devenijn wrote: We are located in Belgium and just ordered a PRA line, the telco asked the following questions : - 120 or 75 ohm ? - Support for CRC4 yes/no ? - B channels in 2 way ? 1) Neither. Digium cards require an RJ-45 connection. Search the mailing list for info on this. I seem to remember seeing talk of many coax (what your telco wants to provide) to RJ-45 converters available. 2) It doesn't matter 3) I have no idea. I assume you always want 2-way. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT - 3com 3C17205 cisco 79xx
Does anyone know if the 3com 3C17025 (which supports NBX phones and IEEE 802.3af ) would work with Cisco 79xx phones for PoE ? Many thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100 or TE410 card an PRA line
Eric Wieling wrote: Michael Devenijn wrote: We are located in Belgium and just ordered a PRA line, the telco asked the following questions : - 120 or 75 ohm ? - Support for CRC4 yes/no ? 120 ohm is an RJ45 connection. YES for the CRC that should be standard for EuroISDN Do not have a clue about the last Theo 1) Neither. Digium cards require an RJ-45 connection. Search the mailing list for info on this. I seem to remember seeing talk of many coax (what your telco wants to provide) to RJ-45 converters available. 2) It doesn't matter 3) I have no idea. I assume you always want 2-way. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Snom 190/220 dialplan strings?
Hi,
Did anyone make sense out of the snom dialplan
strings? I am struggling with it trying to get the
phones to dial 4 digit extensions and 10 digit
numbers without the need for the OK button.
upgrade your phone to 3.56 firmware and
use |([0-9]{4})|sip:[EMAIL PROTECTED]|d to auto dial 4 digit
number.
may be the following post will be useful:
http://lists.digium.com/pipermail/asterisk-users/2004-October/070037.html
Arsen.
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[Asterisk-Users] RE: Setup/SIP routing
I'm still stuck this/my problem.
Even if I create a friend entry and register my softphone directly to
Asterisk, the Dail(${EXTEN},entity) seems to replace the From: part with the
From: 349525 sip:[EMAIL PROTECTED]:5065 part instead of the
From: 349525 sip:[EMAIL PROTECTED]
So if I would add that incoming call to my addressbook the sip URI is wrong.
I'm thinking something in the way of fromuser=${SIPCALLID} would be needed
for this?
I'm also not able to get Asterisk out of the mediastream, I've set the
canreinvite options to yes, but still asterisk stays in the stream.
I've made a SIP scenario trace of the callsetup, I'm a bit puzzled by the 2
time call setup?
Kind regards,
E. Versaevel
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens E. Versaevel
Verzonden: donderdag 18 november 2004 13:53
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] RE: Setup/SIP routing
The problem is that that should be dynamic :/
Take a look at this sip msg:
INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0
Max-Forwards: 10
Record-Route: sip:[EMAIL PROTECTED];ftag=as3f718642;lr=on
Via: SIP/2.0/UDP ser.box;branch=z9hG4bK93dc.71ad80b3.0
Via: SIP/2.0/UDP ser.box:5065;branch=z9hG4bK513b584d
From: 349525 sip:[EMAIL PROTECTED]:5065;tag=as3f718642
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]:5065
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 18 Nov 2004 12:46:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 220
P-hint: USRLOC
v=0
o=root 26383 26383 IN IP4 ser.box
s=session
c=IN IP4 ser.box
t=0 0
m=audio 14682 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
As you can see the from user is not correct, this should be
[EMAIL PROTECTED] If a user adds this entry to a phonebook, the contact
info will be wrong.
-Oorspronkelijk bericht-
Van: Benjamin on Asterisk Mailing Lists
[mailto:[EMAIL PROTECTED]
Verzonden: donderdag 18 november 2004 11:41
Aan: E. Versaevel
Onderwerp: Re: [Asterisk-Users] Setup/SIP routing
Hi
On Thu, 18 Nov 2004 11:32:08 +0100, E. Versaevel [EMAIL PROTECTED] wrote:
However, I'm having troubles routing incoming sip traffic to SER,
asterisks
keeps messing up the form header (replacing it by the dialed context, ie
[EMAIL PROTECTED] )
You can control what Asterisk puts into the FROM header through the
parameters fromuser and fromdomain in sip.conf.
regards
benjamin
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
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IpTel Asterisk
SER
SoftPhone
| |
| |
CallPFrameTime
| |
| |
|F1 INVITE (sdp)-|
| | 1
PF:70 09:27:15.
| |
| |
|-- Trying 100 F2|
| | 1
PF:71 09:27:15.7783
| |
| |
| |F3 INVITE
(sdp)-|
| 2 PF:72 09:27:15.7789
| |
| |
| |-- trying -- your call is
important to us 100 F4| | 2
PF:73 09:27:15.7795
| |
| |
Re: [Asterisk-Users] OT - 3com 3C17205 cisco 79xx
Asterisk wrote: Does anyone know if the 3com 3C17025 (which supports NBX phones and IEEE 802.3af ) would work with Cisco 79xx phones for PoE ? Many thanks. I very much doubt it. I bought a 4400 PWR to test with our Siemens Optipoint handsets, which also support 802.3af. The two do not work together. The official response I was given was that it is due to different detection algorithms in use by the various vendors. In the end we purchased some PowerDsine kit to supply the power, which is supposed to be the most compatible with a range of vendors 8023.af kit. Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] problem with zyxel prestige 2002
Title: R: [Asterisk-Users] problem with zyxel prestige 2002 We also had this problem with the zyxel 2002. Upgrade to the latest firmware, then it will work. Older firmwares had trouble with incoming calls behind NAT. -Manuel -Messaggio originale- Da: Stig Thune [mailto:[EMAIL PROTECTED]] Inviato: lunedì, 15. novembre 2004 19:16 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] problem with zyxel prestige 2002 This sounds odd. We use the same adapter. I will check this more.. Are u sure you have set the phone up correctly ? And also - have to checked the ring phone1 or phone2 on incomming calls ? / Stig Henning - Original Message - From: Mihkel Raba [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 14, 2004 9:51 PM Subject: [Asterisk-Users] problem with zyxel prestige 2002 Hi I tried to use Zyxel Prestige 2002 VoIP Analog Telephone Adapter with asterisk. Device registers both phones and i can call out. But incoming calls are not working. Asterisk - sip show peers shows zyxel, zyxel web interfce shows that devices are registered. But when i do incoming call to zyxel, phones do not ring and if voicemail is configured, calls go directly to voicemail. Any suggestions ? Mihkel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR and voice mail using G729
On Wed, 2004-11-17 at 18:32, Alvaro Gonzalez wrote: I need to know if it is possible to use the IVR and Voicemail using G729, I have two SIP phones that uses G729 and I can not heard the IVR and the voice mail. Yes, you just need to purchase a G.729 licence from Digium. 2 phones, $20 (Presuming 1 channel at a time). Not a bad deal: http://www.digium.com/index.php?menu=asterisk_g729 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unpredictables Hangups
Hello all, i'm experiencing a list of unpredictables hangup on SIP phones using a PRI E100P Card. All i can see in logs is WARNING[81931]: File chan_zap.c, Line 5402 (zt_pri_error): PRI: Read on 37 failed: Unknown error 500 I receive a lot of these errors in asterisk/messages. It doesn't seem to be strictly linked to hangups, since i have dozen of these messages per minute and i have completely random hangups on lines. Moreover, lines don't fall togheter (eg i don't see channels restart nor all phone calls are hungup in the same moment). I'm using stable asterisk 1.0, with correspondant libpri and zaptel, and all phones are Snom 105 SIP Phones. Hangups happen only if i use Zap lines, when i use phones for internal calls all goes fine. Hardware is a P4 1.7Ghz with 512Mb of RAM, on a Gygabite Motherboard. E100P has it's own IRQ level (not shared and below 15). Installation has been done on a RedHat9 system with 2.4 kernel taken from kernel.org (2.4.18). I desperately need any kind of possible help, since here my boss is planning to wipe asterisk from our office and return to a traditional PBX. I can exclude problems in PRI line from Telco and problems due to the cable (it's a shielded cable made from a telco operator) Thanks in advance, -- Stefano Finetti Technical Coordinator Lynx Autodelta S.r.l. Tel.: 199797930 Fax.: 06233227934 email: [EMAIL PROTECTED] -- Outgoing mail is certified Virus Free. Checked by AVG Anti-Virus (http://www.grisoft.com). Version: 7.0.279 / Virus Database: 265.4.0 - Release Date: 18/11/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is H323 dying?
kido noagbodji wrote: Hello, I just downloaded and installed the latest version of asterisk under Fedora. (had it under FreeBSD but was having TOOO many problems) After my installation i noticed that the channel H323 was not included ( I remember that i did not have to install it under freeBSD) but I have seen that SIP and IAX are supported though. So i am wondering: Does asterisk consider H323 so achaic that it does not bother including it anymore? According to you specialists, are we looking at the end of H323? or maybe i just did not install asterisk properly :-). H.323 support for Asterisk based on the original code (asterisk-oh323) is far from dying. Check: http://www.inaccessnetworks.com/projects/asterisk-oh323 for the latest code. Thanks Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P and Siemens Gigaset 4175
Hi I have read on the list about various problems with the the X100P and have tried some of the suggestions but still have problems. I am using the X100P to connect to a Siemens Gigaset 4175 which is an ISDN PBX with DECT extensions. The Gigaset also has two POTS ports and I am trying to connect to one of these. I am based in the UK but bought the Gigaset from Germany. The problem I have is that Asterisk does not always answer or see the incoming calls from the Gigaset. I ring the POTS port from another extension whilst looking at the asterisk console. Sometime it sees the call straightaway and everything goes OK. Sometime it picks up the call after 10 or 20 seconds and sometimes it never sees the call at all. It appears to function well with outgoing calls and I can route SIP calls through to extensions or onto the external ISDN line. It would be quite a neat setup if I can get it to work as I can pass SIP and IAX calls to and from DECT extensions. (DECT is European cordless phone standard) The Gigaset uses US tones (I think) and appears to pass through US callerid to the POTS ports - when it works. Thanks Ian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: Incorrect parsing of 'unavailable' caller-ID fromCisco gateway
Before I raise this as a bug, it appears that * incorrect sets and reads the caller-id field from incoming sip packets when a Cisco gateway doesnt send one. Actually, dug into this further, and its an issue with reading Remote-Party-ID headers from the Cisco in get_rpid_num, so I've raised a bug. http://bugs.digium.com/bug_view_page.php?bug_id=0002910 Thanks for the response, Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Little off topic
I'm going to get 2 T100P cards. One for our Asterisk server and one for the HylaFax Server. Will this work? My next question is can I have Asterisk detect fax tone and route the call to an extension. You call 555-1212 and it's a voice call it goes to his SIP phone. If it's a fax route call to 555-1213. Thanks for your great help -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, November 19, 2004 4:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Little off topic Martin List-Petersen wrote: You can't, the T100P is a unchannelized T1 card. This is 100% wrong. The T100P supports Channelized Voice T-1 (aka CT1) If you want to use it with HylaFax you need either SpanDSP OR an analog port on Asterisk in addition to the T100P. A search of the mailing lists would have told you this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unpredictables Hangups
Google: Results 1 - 10 of about 149 from lists.digium.com for Unknown error 500. Specifically: http://lists.digium.com/pipermail/asterisk-users/2004-April/042912.html http://lists.digium.com/pipermail/asterisk-users/2003-November/028105.html http://lists.digium.com/pipermail/asterisk-users/2003-November/028105.html http://lists.digium.com/pipermail/asterisk-users/2004-April/043860.html http://lists.digium.com/pipermail/asterisk-users/2003-July/016538.html http://lists.digium.com/pipermail/asterisk-users/2004-June/049410.html http://lists.digium.com/pipermail/asterisk-users/2004-July/055478.html http://lists.digium.com/pipermail/asterisk-users/2004-March/040502.html Usually I require dinner and drinks before this kind of heavy duty handholding. Today I guess I was just feeling sorry for someone that can't google, so I figured I'd just give you a freebie. Stefano Finetti wrote: Hello all, i'm experiencing a list of unpredictables hangup on SIP phones using a PRI E100P Card. All i can see in logs is WARNING[81931]: File chan_zap.c, Line 5402 (zt_pri_error): PRI: Read on 37 failed: Unknown error 500 I receive a lot of these errors in asterisk/messages. It doesn't seem to be strictly linked to hangups, since i have dozen of these messages per minute and i have completely random hangups on lines. Moreover, lines don't fall togheter (eg i don't see channels restart nor all phone calls are hungup in the same moment). I'm using stable asterisk 1.0, with correspondant libpri and zaptel, and all phones are Snom 105 SIP Phones. Hangups happen only if i use Zap lines, when i use phones for internal calls all goes fine. Hardware is a P4 1.7Ghz with 512Mb of RAM, on a Gygabite Motherboard. E100P has it's own IRQ level (not shared and below 15). Installation has been done on a RedHat9 system with 2.4 kernel taken from kernel.org (2.4.18). I desperately need any kind of possible help, since here my boss is planning to wipe asterisk from our office and return to a traditional PBX. I can exclude problems in PRI line from Telco and problems due to the cable (it's a shielded cable made from a telco operator) Thanks in advance, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100 or TE410 card an PRA line
Both CRC4 off and CRC4 on will work fine with those cards. Since its a belgian carrier, i probably already set it up in the past. so if needed i could do it again :) zoa. Theodoros Georgiou wrote: Eric Wieling wrote: Michael Devenijn wrote: We are located in Belgium and just ordered a PRA line, the telco asked the following questions : - 120 or 75 ohm ? - Support for CRC4 yes/no ? 120 ohm is an RJ45 connection. YES for the CRC that should be standard for EuroISDN Do not have a clue about the last Theo 1) Neither. Digium cards require an RJ-45 connection. Search the mailing list for info on this. I seem to remember seeing talk of many coax (what your telco wants to provide) to RJ-45 converters available. 2) It doesn't matter 3) I have no idea. I assume you always want 2-way. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internet bandwidth (comparing overhead)
Hi, I like to know why iLBC and GSM generate a 40-50kbps bandwidth Is very high, if compared with your calculations for other codecs(G723.1 / 17kbps and G729 / 24 Kbps). Alexandre Kanuri, Seshu (Company IT) wrote: /SNIP/ Some corrections are needed: 6.3kbps of G723.1 will become around 17kbps on the IP level without silence suppression because of the additional overhead imposed by protocols like RTP, IP, etc . If you add the Ethernet (or WAN protocol overhead) this will increase even more (although slightly). Similarly, a voice stream of G729 at 8kbps will become around 24kbps on the IP level, and slightly more on the Ethernet or ppp level (around 25 kbps). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need help selecting phones
Im new to the asterisk world and have been playing with an asterisk server with 1 FXO card for a couple of weeks. Now Im looking to start adding IP Desk Phones but Im unable to come to a decision on what phones to use. I like to look of the Polycoms, but because we are not a phone company I cant see us getting reseller authorized for them. Shoretel has some nice looking phones, but I dont want to be forced into buys their PBXs as well. I dont like to look of the grandstream budgetel stuff as it looks like its name implies. I would really like SIP, multi-line display, multiple extensions, and handsfree. Can someone recommend a line of phones that work well with * and are distributed in Canada? Id prefer a distributor located in Canada but that is not a priority over getting a good line of phones that I fell we can put side by side any digital system and say we can do everything that phone can do. Any recommendations would be great. Thanks Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interrupting MusicOnHold while call in queue ?
You can either do a make samples (which will overwrite your existing configuration) or look in the configs subdirectory in the asterisk source tree for queues.conf.sample. It would be nicer if make samples would install the configs as .sample, for example: /etc/asterisk/iax.conf.sample. That way it's safe to do make samples (which is always a good thing) and it's easy to add changes because all you have to do is add these *.sample files to CVS and do a cvs diff -u after you've installed a new version. That way new lines and changes in defaults are spotted and can be put in the live configuration files. Might check to be sure, but I'm thinking their was code in the make file to not over-write matching file names (or was that in zaptel). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internet bandwidth (comparing overhead)
For real life bandwidth tests : check the ppt on www.astertest.com Zoa. alexandre::aldeia digital wrote: Hi, I like to know why iLBC and GSM generate a 40-50kbps bandwidth Is very high, if compared with your calculations for other codecs(G723.1 / 17kbps and G729 / 24 Kbps). Alexandre Kanuri, Seshu (Company IT) wrote: /SNIP/ Some corrections are needed: 6.3kbps of G723.1 will become around 17kbps on the IP level without silence suppression because of the additional overhead imposed by protocols like RTP, IP, etc . If you add the Ethernet (or WAN protocol overhead) this will increase even more (although slightly). Similarly, a voice stream of G729 at 8kbps will become around 24kbps on the IP level, and slightly more on the Ethernet or ppp level (around 25 kbps). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: MARIO SPOLJAR is not longer working for PLIVA
Just got the message below from the Pliva people. Does someone have admin access to the list to remove him? Begin forwarded message: From: Modric, Kristijan [EMAIL PROTECTED] Date: November 19, 2004 6:57:47 AM EST To: [EMAIL PROTECTED] Subject: RE: MARIO SPOLJAR is not longer working for PLIVA Hi, MARIO SPOLJAR is no longer PLIVA´s employee. Can you please remov him from that list. Thanks! Best Regards, Kristijan Modric -Original Message- From: Popov, Dragica Sent: Friday, November 19, 2004 8:44 AM To: Modric, Kristijan Subject: FW: MARIO SPOLJAR is not longer working for PLIVA -Original Message- From: Noah Miller [mailto:[EMAIL PROTECTED] Sent: Thursday, November 18, 2004 3:59 PM To: Administrator Subject: Re: MARIO SPOLJAR is not longer working for PLIVA Please adjust your autoreply settings. Every time we post to a list to which MARIO SPOLJAR was subscribed, EVERYONE on the list gets a message that he is no longer employed. A mail server should only bounce messages in which the addressed party is directly referenced in a To: or Cc: field. On Nov 18, 2004, at 9:54 AM, Pliva d.d.postmaster wrote: Primatelj va¹eg e-maila: MARIO SPOLJAR vi¹e nije zaposlenik PLIVE --- The recipient: MARIO SPOLJAR is no longer PLIVA´s employee. This e-mail, and any attachments, may contain confidential, and/or legally privileged information, and is intended only to be seen and used by the named addressee(s). If you have received this e-mail in error, please notify the sender immediately, and permanently delete the original and any copies of the e-mail, and any attachments, without printing, reading or copying them. Any use, distribution, or copying of this e-mail other than by the intended recipient is strictly prohibited. PLIVA accepts no liability for the content of this email, or for the consequences of any actions taken on the basis of the information provided. Thank you for your co-operation. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog ports via USB
RE: the S100Us - I think you can get them from www.tjnet.com (TigerJet). You are probably after their USB to RJ11 adapter. I think that the Personal Phone Gateway-PCI cards are generic X100Ps too (they look identical except no heat sink glued to the chip). I'm guessing that TigerJet supply the POTS hardware to digium? They allow you to buy some samples from their Yahoo shop. Derek Martin List-Petersen wrote: On Thu, 2004-11-18 at 14:03, Ed Greenberg wrote: Over on the voip-info.org tiki I found this statement: Mark (the man who made Asterisk PBX, www.asterisk.org) has an xbox that has 4 analog ports via usb... aka the XBoxPBX While I'm not interested in the xbox part of this, I wonder how one uses USB for analog connections? Explanation? Pointer to an article? Other info? zaptel .. just as regular. There are wcusb.o modules in the zaptel drivers, that handle these. The usb fxs modules are part of the DevKit lite, that Asterisk was selling. I can however not see them anymore on the site. Slán leat, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 for *
I just avoid people who think it's ok to create proprietary extensions to free software. People like that should be ashamed of themselves, as it's just an insult to the people who have freely contributed to the project. I fully agree. How hard would it be to integrate OpenSS7.org with Asterisk and use a Cisco IPT Signalling point to terminate the A-Links? A lot of the puzzle pieces exist, they just need to be plugged together. -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Routing between different interfaces
I have an asterisk box with a public IP for people on the Internet to connect to. I also have a Lucent TNT on the same physical network but on a 10.0.0.0 subnet. It isn't safe to put the TNT on a public IP address and I never want it to talk to the net directly anyhow so this seemed like a good idea. However asterisk does not seem to properly route SIP calls between the interfaces. I tell the TNT to only allow connections from the ip of the asterisk box but the IP in the SIP headers comes through as that of the originating box, not the asterisk box. Is this how it is supposed to work? It would seem to make impossible what I want to do. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgpOySRSM9AKZ.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 for *
Matthew Crocker wrote: I just avoid people who think it's ok to create proprietary extensions to free software. People like that should be ashamed of themselves, as it's just an insult to the people who have freely contributed to the project. I fully agree. How hard would it be to integrate OpenSS7.org with Asterisk and use a Cisco IPT Signalling point to terminate the A-Links? A lot of the puzzle pieces exist, they just need to be plugged together. The reason we built a new SS7 stack from scratch was we looked at openss7 :-) I can't imagine anyone successfully integrating openss7 into anything. I believe it works OK on its own, and is in use as a gateway. It wasn't designed to play nicely with anyone else, though. There have been a number of projects trying to use openss7 as a part of something else, including multiple efforts to make it work with *. I haven't heard of any of any of them succeeding. We paid the US$1k you need to pay to get access to the openss7 code, and it just wasted our time. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_sms: problems sending a sms
Hello,
i try to send out a sms, but with no success.
The trunk is a E100P, and the sms should go out to the
Telekom SM-SC. What i want to to at the first run is,
sending out a sms when a certain number is dialed.
I tried:
In extensions.conf:
exten = 35953,1,SMS(${TRUNK}/9350193010,,0179NUMBER,Hi there)
exten = 35953,2,SMS(${TRUNK}/9350193010)
exten = 35953,3,Hangup
exten = 35954,1,Dial(${TRUNK}/9350193010)
and get:
tkserv*CLI
-- Executing Goto(SIP/35903-da57, voiplocal|35953|1) in new stack
-- Goto (voiplocal,35953,1)
-- Executing SMS(SIP/35903-da57, Zap/g1/9350193010||0179NUMBER|Hi
there) in new stack
-- Executing SMS(SIP/35903-da57, Zap/g1/9350193010) in new stack
-- SMS TX 92 01 FF 6E 00 00...
-- Executing Hangup(SIP/35903-da57, ) in new stack
== Spawn extension (voiplocal, 35953, 3) exited non-zero on 'SIP/35903-da57'
935 is the prefix to go out to the world via a telekom PRI line.
Sometimes i hear a chirp like the sound of a bird, sometimes i
get this SMS TX 92 01 FF 6E 00 00... line, sometimes nothing
happens but a hangup after a few seconds.
(0179NUMBER is the number of the cell-phone).
When i call the 35954 via a SIP Phone, i hear always one chirp,
and a hangup after a few seconds, so i guess the call reaches
the SM-SC.
Does someone know whats wrong?
cu,
Steffen
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RE: [Asterisk-Users] Routing between different interfaces
canreinvite=no ? http://www.voip-info.org/wiki-Asterisk+sip+canreinvite -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tracy R Reed Sent: Friday, November 19, 2004 6:56 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Routing between different interfaces I have an asterisk box with a public IP for people on the Internet to connect to. I also have a Lucent TNT on the same physical network but on a 10.0.0.0 subnet. It isn't safe to put the TNT on a public IP address and I never want it to talk to the net directly anyhow so this seemed like a good idea. However asterisk does not seem to properly route SIP calls between the interfaces. I tell the TNT to only allow connections from the ip of the asterisk box but the IP in the SIP headers comes through as that of the originating box, not the asterisk box. Is this how it is supposed to work? It would seem to make impossible what I want to do. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] rtp codec error
Hello all, I just register my asterisk with Digium g729 codec. But now when I place a call with my SIP phone through my Cisco ATA 186 box, I have this error: rtp.c:319 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256. Can some body tell me why?? Part of my SIP.conf: Disallow=all Allow=g729 Allow=alaw [5000500] type=friend callgroup=1 host=dynamic defaultip=xxx.xxx.xxx.xxx dtmfmode=rfc2833 context=sip-provider allow=g729 allow=alaw canreinvite=no callerid=John 5000500 mailbox=5000500 pickupgroup=1 Thanks. image002.jpg___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT - 3com 3C17205 cisco 79xx
Cisco 79xx phones are NOT 802.3af compliant or even compatible. If you have a mid-span 802.3af injector, this can work with the phone, provide you follow the instructions at - http://www.voip-info.org/tiki-index.php?page=Cisco%20POE If you have an end-span injector, such as 3-com switch forget it, it will never work. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Friday, 19 November 2004 5:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT - 3com 3C17205 cisco 79xx Does anyone know if the 3com 3C17025 (which supports NBX phones and IEEE 802.3af ) would work with Cisco 79xx phones for PoE ? Many thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Routing between different interfaces
On Fri, Nov 19, 2004 at 07:44:34AM -0600, Tim Jackson spake thusly: canreinvite=no ? I already thought of that and canreinvite is already set to no. I also know about bindaddr and localnet but neither of those do what I want either. Thanks. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgpLpyltJAHEh.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unpredictables Hangups
From: Eric Wieling [EMAIL PROTECTED] Google: Results 1 - 10 of about 149 from lists.digium.com for Unknown error 500. Specifically: http://lists.digium.com/pipermail/asterisk-users/2004-April/042912.html http://lists.digium.com/pipermail/asterisk-users/2003-November/028105.html http://lists.digium.com/pipermail/asterisk-users/2003-November/028105.html http://lists.digium.com/pipermail/asterisk-users/2004-April/043860.html http://lists.digium.com/pipermail/asterisk-users/2003-July/016538.html http://lists.digium.com/pipermail/asterisk-users/2004-June/049410.html http://lists.digium.com/pipermail/asterisk-users/2004-July/055478.html http://lists.digium.com/pipermail/asterisk-users/2004-March/040502.html Usually I require dinner and drinks before this kind of heavy duty handholding. Today I guess I was just feeling sorry for someone that can't google, so I figured I'd just give you a freebie. Well Eric... The fact is that i googled a lot. Read ALL the post you just linked, and a lot more. I've tried almost all that solutions... And if you had followed other pages on the google result you should have seen a thread opened by myself about 1 year ago on the list... Actually, i am in this situation: E100P with NO shared IRQs: [EMAIL PROTECTED] /root]# cat /proc/interrupts CPU0 0: 27710902 XT-PIC timer 1: 3 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 39199525 XT-PIC eth0 10: 276808484 XT-PIC t1xxp 11:4083904 XT-PIC Cyclades-PC300 12: 0 XT-PIC PS/2 Mouse 14: 237800 XT-PIC ide0 15: 3 XT-PIC ide1 NMI: 0 LOC: 27711957 ERR: 0 No USB enabled on Motherboard, IRQ fixed via Motherboard BIOS in order to be absolutely sure that it won't be shared with anything (as the /proc/interrupts output shows well) I still go Read on 32 and Read on 37 fails with unknown error 500. Using Zttool, i see some Lost Interrupt, but even if i reduced a lot the number of irq losses, they still there. It seems that there is no way to completely stop irq losses, but the problem is that actually i can't do anything to stop irq losses, because i ran out of ideas on how to solve this problem, and this is the reason why i was asking again here, in the hope of someone who found the same problem recently and found an appropriate solution. Following one of the advices from this thread, i checked for busydetect and busycount. Now i'm monitoring if calls are still dropped randomly. But it seems that PRI error and Call drops are not so-strictly linked... Thanks anyway for the help :-) -- Stefano Finetti Technical Coordinator Lynx Autodelta S.r.l. Tel.: 199797930 Fax.: 06233227934 email: [EMAIL PROTECTED] -- Outgoing mail is certified Virus Free. Checked by AVG Anti-Virus (http://www.grisoft.com). Version: 7.0.279 / Virus Database: 265.4.0 - Release Date: 18/11/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog ports via USB
Derek Conniffe schrieb: Re: the S100Us - I think you can get them from www.tjnet.com (TigerJet). You are probably after their USB to RJ11 adapter. I think that the Personal Phone Gateway-PCI cards are generic X100Ps too Do you know if the USB phone and the USB IP Phone adaptor is Linux compatible? Bye! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unpredictables Hangups
Stefano Finetti wrote: From: Eric Wieling [EMAIL PROTECTED] Google: Results 1 - 10 of about 149 from lists.digium.com for Unknown error 500. Specifically: http://lists.digium.com/pipermail/asterisk-users/2004-April/042912.html http://lists.digium.com/pipermail/asterisk-users/2003-November/028105.html http://lists.digium.com/pipermail/asterisk-users/2003-November/028105.html http://lists.digium.com/pipermail/asterisk-users/2004-April/043860.html http://lists.digium.com/pipermail/asterisk-users/2003-July/016538.html http://lists.digium.com/pipermail/asterisk-users/2004-June/049410.html http://lists.digium.com/pipermail/asterisk-users/2004-July/055478.html http://lists.digium.com/pipermail/asterisk-users/2004-March/040502.html Usually I require dinner and drinks before this kind of heavy duty handholding. Today I guess I was just feeling sorry for someone that can't google, so I figured I'd just give you a freebie. Well Eric... The fact is that i googled a lot. Read ALL the post you just linked, and a lot more. I've tried almost all that solutions... And if you had followed other pages on the google result you should have seen a thread opened by myself about 1 year ago on the list... Actually, i am in this situation: E100P with NO shared IRQs: [EMAIL PROTECTED] /root]# cat /proc/interrupts CPU0 0: 27710902 XT-PIC timer 1: 3 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 39199525 XT-PIC eth0 10: 276808484 XT-PIC t1xxp 11:4083904 XT-PIC Cyclades-PC300 12: 0 XT-PIC PS/2 Mouse 14: 237800 XT-PIC ide0 15: 3 XT-PIC ide1 NMI: 0 LOC: 27711957 ERR: 0 No USB enabled on Motherboard, IRQ fixed via Motherboard BIOS in order to be absolutely sure that it won't be shared with anything (as the /proc/interrupts output shows well) I still go Read on 32 and Read on 37 fails with unknown error 500. Using Zttool, i see some Lost Interrupt, but even if i reduced a lot the number of irq losses, they still there. It seems that there is no way to completely stop irq losses, but the problem is that actually i can't do anything to stop irq losses, because i ran out of ideas on how to solve this problem, and this is the reason why i was asking again here, in the hope of someone who found the same problem recently and found an appropriate solution. Following one of the advices from this thread, i checked for busydetect and busycount. Now i'm monitoring if calls are still dropped randomly. But it seems that PRI error and Call drops are not so-strictly linked... Did you mention your extensive google search in your message? If so I didn't notice it. Do you have a /etc/sysconfig/harddisks ? If so uncomment USE_DMA=1, MULTIPLE_IO=16, EIDE_32BIT=3, LOOKAHEAD=1 and add -u1 to EXTRA_PARAMS. If you already have those enabled, try commenting out all of them except the EXTRA_PARAMS=-u1 If all else fails replace the motherboard with something different (different brand/chipset/etc). Are you using any RAID? If so disable it. Several people have reported problems with Promise RAID that were solved when they removed the Promise RAID card. --Eric ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] helo
Helo test brazil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unpredictables Hangups
Eric, What state are you in? Ron Original Message From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Unpredictables Hangups Date: Fri, 19 Nov 2004 08:10:32 -0600 Stefano Finetti wrote: From: Eric Wieling [EMAIL PROTECTED] Google: Results 1 - 10 of about 149 from lists.digium.com for Unknown error 500. Specifically: http://lists.digium.com/pipermail/asterisk-users/2004-April/042912.ht ml http://lists.digium.com/pipermail/asterisk-users/2003-November/028105 .html http://lists.digium.com/pipermail/asterisk-users/2003-November/028105 .html http://lists.digium.com/pipermail/asterisk-users/2004-April/043860.ht ml http://lists.digium.com/pipermail/asterisk-users/2003-July/016538.htm l http://lists.digium.com/pipermail/asterisk-users/2004-June/049410.htm l http://lists.digium.com/pipermail/asterisk-users/2004-July/055478.htm l http://lists.digium.com/pipermail/asterisk-users/2004-March/040502.ht ml Usually I require dinner and drinks before this kind of heavy duty handholding. Today I guess I was just feeling sorry for someone that can't google, so I figured I'd just give you a freebie. Well Eric... The fact is that i googled a lot. Read ALL the post you just linked, and a lot more. I've tried almost all that solutions... And if you had followed other pages on the google result you should have seen a thread opened by myself about 1 year ago on the list... Actually, i am in this situation: E100P with NO shared IRQs: [EMAIL PROTECTED] /root]# cat /proc/interrupts CPU0 0: 27710902 XT-PIC timer 1: 3 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 39199525 XT-PIC eth0 10: 276808484 XT-PIC t1xxp 11:4083904 XT-PIC Cyclades-PC300 12: 0 XT-PIC PS/2 Mouse 14: 237800 XT-PIC ide0 15: 3 XT-PIC ide1 NMI: 0 LOC: 27711957 ERR: 0 No USB enabled on Motherboard, IRQ fixed via Motherboard BIOS in order to be absolutely sure that it won't be shared with anything (as the /proc/interrupts output shows well) I still go Read on 32 and Read on 37 fails with unknown error 500. Using Zttool, i see some Lost Interrupt, but even if i reduced a lot the number of irq losses, they still there. It seems that there is no way to completely stop irq losses, but the problem is that actually i can't do anything to stop irq losses, because i ran out of ideas on how to solve this problem, and this is the reason why i was asking again here, in the hope of someone who found the same problem recently and found an appropriate solution. Following one of the advices from this thread, i checked for busydetect and busycount. Now i'm monitoring if calls are still dropped randomly. But it seems that PRI error and Call drops are not so-strictly linked... Did you mention your extensive google search in your message? If so I didn't notice it. Do you have a /etc/sysconfig/harddisks ? If so uncomment USE_DMA=1, MULTIPLE_IO=16, EIDE_32BIT=3, LOOKAHEAD=1 and add -u1 to EXTRA_PARAMS. If you already have those enabled, try commenting out all of them except the EXTRA_PARAMS=-u1 If all else fails replace the motherboard with something different (different brand/chipset/etc). Are you using any RAID? If so disable it. Several people have reported problems with Promise RAID that were solved when they removed the Promise RAID card. --Eric ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best line protocol for T1
Hello all, I'm provisioning a T1-PRI for a Digium T410P with my local TELCO.The TELCO has asked me to picka line protocol and has theoption of several RBS protocols, like 5ESS (Lucent), IN2 and others. The switch is a 5ESS, but the "normal" (according to the sales rep) protocol is IN2. I see from the doc on zaptel, that many protocols are supported. As I have a choice, is one of the PRI protocols "better" that an another? I suspect that some protocols support features over another but can't find much on the specifics. Recommendations and reasons please. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Little off topic
Citat Eric Wieling [EMAIL PROTECTED]: Martin List-Petersen wrote: You can't, the T100P is a unchannelized T1 card. This is 100% wrong. The T100P supports Channelized Voice T-1 (aka CT1) If you want to use it with HylaFax you need either SpanDSP OR an analog port on Asterisk in addition to the T100P. Might be that i'm wrong on the unchannelized bit, but i don't see, where the analog port will help you ? The guy wants to do Hylafax directly on a T100P w/o Asterisk or Asterisk as middleware, which i don't see working. SpanDSP on the other side works well, but that is basically a softmodem emulation, something Hylafax can't do. I have not seen any applications for spandsp outside Asterisk, yet. Slán Leat, Martin List-Petersen Dublin, Eire (contact info == http://www.marlow.dk) -- linux: because a PC is a terrible thing to waste ([EMAIL PROTECTED] put this on Tshirts in '93) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hello
New user * Test Brasil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unpredictables Hangups
Near New Orleans Louisiana, but I am interested in long term, part time consulting work in the Toronto, ON area. [EMAIL PROTECTED] wrote: Eric, What state are you in? Ron Original Message From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Unpredictables Hangups Date: Fri, 19 Nov 2004 08:10:32 -0600 Stefano Finetti wrote: From: Eric Wieling [EMAIL PROTECTED] Google: Results 1 - 10 of about 149 from lists.digium.com for Unknown error 500. Specifically: http://lists.digium.com/pipermail/asterisk-users/2004-April/042912.ht ml http://lists.digium.com/pipermail/asterisk-users/2003-November/028105 .html http://lists.digium.com/pipermail/asterisk-users/2003-November/028105 .html http://lists.digium.com/pipermail/asterisk-users/2004-April/043860.ht ml http://lists.digium.com/pipermail/asterisk-users/2003-July/016538.htm l http://lists.digium.com/pipermail/asterisk-users/2004-June/049410.htm l http://lists.digium.com/pipermail/asterisk-users/2004-July/055478.htm l http://lists.digium.com/pipermail/asterisk-users/2004-March/040502.ht ml Usually I require dinner and drinks before this kind of heavy duty handholding. Today I guess I was just feeling sorry for someone that can't google, so I figured I'd just give you a freebie. Well Eric... The fact is that i googled a lot. Read ALL the post you just linked, and a lot more. I've tried almost all that solutions... And if you had followed other pages on the google result you should have seen a thread opened by myself about 1 year ago on the list... Actually, i am in this situation: E100P with NO shared IRQs: [EMAIL PROTECTED] /root]# cat /proc/interrupts CPU0 0: 27710902 XT-PIC timer 1: 3 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 39199525 XT-PIC eth0 10: 276808484 XT-PIC t1xxp 11:4083904 XT-PIC Cyclades-PC300 12: 0 XT-PIC PS/2 Mouse 14: 237800 XT-PIC ide0 15: 3 XT-PIC ide1 NMI: 0 LOC: 27711957 ERR: 0 No USB enabled on Motherboard, IRQ fixed via Motherboard BIOS in order to be absolutely sure that it won't be shared with anything (as the /proc/interrupts output shows well) I still go Read on 32 and Read on 37 fails with unknown error 500. Using Zttool, i see some Lost Interrupt, but even if i reduced a lot the number of irq losses, they still there. It seems that there is no way to completely stop irq losses, but the problem is that actually i can't do anything to stop irq losses, because i ran out of ideas on how to solve this problem, and this is the reason why i was asking again here, in the hope of someone who found the same problem recently and found an appropriate solution. Following one of the advices from this thread, i checked for busydetect and busycount. Now i'm monitoring if calls are still dropped randomly. But it seems that PRI error and Call drops are not so-strictly linked... Did you mention your extensive google search in your message? If so I didn't notice it. Do you have a /etc/sysconfig/harddisks ? If so uncomment USE_DMA=1, MULTIPLE_IO=16, EIDE_32BIT=3, LOOKAHEAD=1 and add -u1 to EXTRA_PARAMS. If you already have those enabled, try commenting out all of them except the EXTRA_PARAMS=-u1 If all else fails replace the motherboard with something different (different brand/chipset/etc). Are you using any RAID? If so disable it. Several people have reported problems with Promise RAID that were solved when they removed the Promise RAID card. --Eric ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fedora Core 3 supported?
Ok I've just joined and attempted to search the archives but have not found anything... Is Fedora Core 3 Supported? Directions for Fedora core 3 install if available would be nice. If not I'll be attempting it anyway and can start a crude set. Assuming that they do not does anybody have a set for Fedora core 2? Not a Linux Beginner (I've done RedHat up to 8.0 for several years now). But a beginner to Fedora, We might in the end move to RedHat ES 3.0 but for initial testing we'd prefer FC3. Thanks ahead of time. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk crashes with Unicall
Hi,
For the last 40 days i've been using Unicall on an Asterisk connected to
an Ericsson MD-110 PBX.
It was working fine for two weeks when there were just some random calls
but for the last two weeks when the load increased to between 5 and 10
simultaneous calls the system became unreliable with 2 main problems:
1- Some dropped calls when the call comes from Unicall: Unicall -
IAX/SIP. When it comes from Zap (E1 PRI) there is no problem: Zap -
IAX/SIP.
2- Asterisk crashes 2 or 3 times a day. Always when there is some
Unicall channel active.
To be sure that the crashes are Unicall related I created an test
enviroment:
2 servers with the same configuration:
- P4 2.8Ghz
- 512MB
- 1 Digium E100P (connected with each other using a E1 cross cable)
The test was: using an .call file to start a call from 1 server to the
other on an extension that dial to the first server, that dial to the
other and so on... until there is no more channels available.
The result: the calls start ringing in both servers until there is no
more channels free, then they start to timeout and hangup. Until here
there is no problem, but then suddenly one of the Asterisk servers
crashes. Sometimes the server that initiated the calls, sometimes the
other, there is no pattern. (I repeated the test several times and one
time both Asterisk crashed).
If I change the signalling to E1 PRI and make the same test there is no
problem (calls ring until no more channels are available and timeout
after some time).
Some messages from the Asterisk that crashed follows below (Got only the
last 200 lines, the complete log is 1800 lines / 192 Kb, too big for
posting here).
Is there some debugging info i can extract from this test and post here
to help ?
Thanks,
Leonardo
zaptel.conf:
span=1,1,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
unicall.conf:
[channels]
language=br
context=principal_in
rxwink=300
usecallerid=yes
hidecallerid=no
usecallingpres=yes
callprogress=no
restrictcid=no
immediate=no
callwaitingcallerid=no
threewaycalling=no
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
protocolclass=mfcr2
protocolvariant=br,4,4
protocolend=cpe #co on the other server
group=1
callerid=asreceived
context=principal_in
channel=1-15
channel=17-31
extensions.conf:
[principal_in]
exten = _.,1,SetCallerId()
exten = _.,2,Dial(UniCall/g1/${EXTEN},600)
Call file:
Channel: UniCall/g1/
Callerid:
MaxRetries: 0
RetryTime: 600
WaitTime: 600
Context: principal_in
Extension: 777
Priority: 1
Core file:
Core was generated by `/usr/sbin/asterisk -fg'.
Program terminated with signal 11, Segmentation fault.
#0 0x407fa684 in ?? ()
No debugging symbols on asterisk binary cause it was installed from the
RPM and the building process strips symbols. I can install other binary
with debugging if it helps...
Asterisk messages:
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Detected
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Making a new call with CRN
32769
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx bits 0xD [2/
2/101/ 0]
Nov 19 09:41:25 WARNING[7175]: UC event Detected
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 on [2/
2/101/ 0]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 on [2/
2/101/ 0]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 off [2/
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 off [2/
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 on [2/
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 on [2/
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 off [2/
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 off [2/
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 on [2/
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 on [2/
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 off [2/
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 off [2/
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 on [2/
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 5 on [2/
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 off [2/
2/102/105]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 5 off [2/
2/102/105]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 6 on [2/
2/102/105]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 5 on [2/
2/102/105]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 6 off [2/
2/102/103]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 5 off [2/
2/102/103]
Nov 19 09:41:26 WARNING[7175]: UniCall: mfcr2 Rx tone 6 on [2/
2/102/103]
Nov 19 09:41:26 WARNING[7175]: UniCall: mfcr2 Tx tone 5 on [2/
2/102/103]
Nov 19 09:41:26 WARNING[7175]: UniCall: mfcr2 Rx tone 6 off [2/
2/102/103]
Nov 19 09:41:26 WARNING[7175]: UniCall: mfcr2 Tx tone 5 off [2/
2/102/103]
Nov 19
Re: [Asterisk-Users] Little off topic
Martin List-Petersen wrote: Citat Eric Wieling [EMAIL PROTECTED]: Martin List-Petersen wrote: You can't, the T100P is a unchannelized T1 card. This is 100% wrong. The T100P supports Channelized Voice T-1 (aka CT1) If you want to use it with HylaFax you need either SpanDSP OR an analog port on Asterisk in addition to the T100P. Might be that i'm wrong on the unchannelized bit, but i don't see, where the analog port will help you ? The guy wants to do Hylafax directly on a T100P w/o Asterisk or Asterisk as middleware, which i don't see working. SpanDSP on the other side works well, but that is basically a softmodem emulation, something Hylafax can't do. I have not seen any applications for spandsp outside Asterisk, yet. *nod* I mist have missed the part about doing it all within Asterisk. I think I wrote that message before my 2nd cup of coffee. An analog port would allow you to plug a modem into the Asterisk box and run Hylafax using that. T-1- Asterisk - Analog - Modem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internet bandwidth (comparing overhead)
On 19/11/2004 21:13 alexandre::aldeia digital said the following: I like to know why iLBC and GSM generate a 40-50kbps bandwidth Is very high, if compared with your calculations for other codecs(G723.1 / 17kbps and G729 / 24 Kbps). the other codecs have better compression, but there's a higher computational price to pay to get that higher compression. for IAX-IAX calls, i've found GSM to be more than adequate, especially with IAX trunking turn on where each additional call just tags on 17kbps in bandwidth. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 for *
On 19/11/2004 21:30 Steve Underwood said the following: I can't imagine anyone successfully integrating openss7 into anything. I believe it works OK on its own, and is in use as a gateway. It wasn't as a gateway between what ? if it's SS7 on one side, what's on the other ? SIGTRAN (SS7 over IP) on top of SCTP ? We paid the US$1k you need to pay to get access to the openss7 code, and it just wasted our time. from my impression of the www.openss7.org site, it looked like they were licensing the source under the GPL, with other bits under the LGPL. does the license you bought specifically for handling closed source uses of the openss7 code ? but seriously, we are interested in the ss7 for * work you've done, and the need for a commercial license doesn't phase us. we don't mind paying for it, but it needs to be asterisk on freebsd. would this be doable ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unpredictables Hangups
Hi again Stefano, I noticed your E100P card generates 10 times as many interupts as your timer - don't know if that could be the issue. On my own system the E110P and two TDM400P cards generates aprox. the same number of interupts as the timer. [EMAIL PROTECTED] root# cat /proc/interrupts CPU0 0: 214498966IO-APIC-edge timer 1: 17IO-APIC-edge i8042 9: 0 IO-APIC-level acpi 12:151IO-APIC-edge i8042 14: 13IO-APIC-edge ide0 17:1180813 IO-APIC-level 3ware Storage Controller 18: 214453475 IO-APIC-level t1xxp 19: 214461664 IO-APIC-level wctdm 20: 214486836 IO-APIC-level wctdm 21: 2 IO-APIC-level fcpcipnp 22: 15138050 IO-APIC-level eth0 23: 906455 IO-APIC-level eth1 NMI: 0 LOC: 214514953 ERR: 0 MIS: 0 [EMAIL PROTECTED] root# A little system-background: Supermicro Mainboard with P4 2.53GHz 2 x Onboard Intel Corp. 82540EM Gigabit Ethernet Controller 512 MB Ram Linux-2.6.8.1 - with apic enabled. Echanced Real Time Clock Support is not compiled into the kernel. Asterisk, Libpri and zaptel are all from the 1.0.2 stable release 3ware runs RAID 5 on 3 disks, with one hot-spare. Hope it can be of any help. Best Regards Michael Actually, i am in this situation: E100P with NO shared IRQs: [EMAIL PROTECTED] /root]# cat /proc/interrupts CPU0 0: 27710902 XT-PIC timer 1: 3 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 39199525 XT-PIC eth0 10: 276808484 XT-PIC t1xxp 11:4083904 XT-PIC Cyclades-PC300 12: 0 XT-PIC PS/2 Mouse 14: 237800 XT-PIC ide0 15: 3 XT-PIC ide1 NMI: 0 LOC: 27711957 ERR: 0 No USB enabled on Motherboard, IRQ fixed via Motherboard BIOS in order to be absolutely sure that it won't be shared with anything (as the /proc/interrupts output shows well) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice
Anybody else having broadvoice problems? -- Executing SetAccount(SIP/101-d03b, LD) in new stack -- Executing Dial(SIP/101-d03b, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 408 Request Timeout back from 147.135.0.128 == No one is available to answer at this time Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 for *
Dinesh Nair wrote: On 19/11/2004 21:30 Steve Underwood said the following: I can't imagine anyone successfully integrating openss7 into anything. I believe it works OK on its own, and is in use as a gateway. It wasn't as a gateway between what ? if it's SS7 on one side, what's on the other ? SIGTRAN (SS7 over IP) on top of SCTP ? Yep, that kind of gateway. He has his own SCTP, and doesn't use the native Linux 2.6 one. We paid the US$1k you need to pay to get access to the openss7 code, and it just wasted our time. from my impression of the www.openss7.org site, it looked like they were licensing the source under the GPL, with other bits under the LGPL. does the license you bought specifically for handling closed source uses of the openss7 code ? Its GPL, but you need a password for CVS, and that costs $1k. Since its GPL, there is nothing to stop you making a mirror, I guess. We didn't know anyone else with a copy, so we paid. As I said, it just wasted our time. but seriously, we are interested in the ss7 for * work you've done, and the need for a commercial license doesn't phase us. we don't mind paying for it, but it needs to be asterisk on freebsd. would this be doable ? I understand people have TE405P running on BSD now. If that is correct, there shouldn't be a lot else to do. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 Status
Hello all, somebody can tell me how h.323 status is? it is working OK?... it has implemented faststart and tunneling per peer based?... thanks a lot!! Sebastian from Argentina. --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.791 / Virus Database: 535 - Release Date: 2004-11-08 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SOLVED: Help wanted getting Busy / Congested working properly
Hi all,
I have Asterisk sat between the PSTN and a PBX. Input and output is E1 PRI
When people from the PSTN call a line on the PBX which is engaged, the
line just sits there silently until they hang up.
It is there in the Wiki, but not where I was looking.
A working way to handle busy/congestion signals in this situation is:
exten = _X.,1,Dial(${PSTN}/${EXTEN})
exten = _X.,2,SetVar(PRI_CAUSE=${HANGUPCAUSE})
exten = _X.,3,Hangup
Dan
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Re: [Asterisk-Users] Re: VOIP security on an IAX connection.
Ditto. There's another very clear advantage to OpenVPN over IPsec, and that's the fact that many firewalls are hard to run IPsec through, but OpenVPN, using a single ephemeral UDP link, will work just fine. I believe that the original poster is not concerned with getting it through a Linksys router at home, and that he has a highi degree of control over which hardware is in the trunk path. I could be wrong, but that's what it sounded like to me. I just tried to get it working last night, and I found it (OpenVPN) no easier as a VPN solution than OpenSWAN was, either in server setup and understanding, or client setup and use. My users and myself are running the XP SP2 and Win2K (updated) MS builtin client into the network through one of those hated Linksys routers, with no problems whatsoever. In the end, I decided that I'd rather stick with the open standards, than wait and hope that the OpenVPN proprietary software became a de-facto standard (isn't that what you all hated Microsoft for? But I digress...) For a single point-to-point link, like the poster requested, with Linux on both ends, there is no reason I can tell to go a proprietary route when IPSec works just fine and comes with the 2.6 kernel (or can be fitted on a 2.4 kernel just fine). Greg Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lobotomized Sipura SPA-3000 configuration needed
I have my SPA-3000 taking a PSTN line inbound and forwarding it to my Asterisk server after a few rings. I don't hear any dial tone when I do that kind of forwarding. I do it via the dial plan and I also tried it via CFwd SelX Caller/Dest. How are you attempting to do it? I am just starting in the configuration of it and didn't get to finish it yesterday; if I get time today I will get back to it with the suggestions in this thread. Thanks! Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel init script
I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having an issue with the zaptel init script.. If I run.. #modprobe zaptel #modprobe wcfxo #modprobe wcfxs .. from a command line it load and appears to be working fine.. If I try and use the init script I get errors about ZT_CHANCONFIG and the modules don't see to laod up.. Anyone got any pointers? I am running Fedora Core 2 with all the updates and I have an X100P and a TDM400P with a single FXS module.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unpredictables Hangups
Michael, I just check'd my kernel configuration... I have APIC support and no Enhanced Real Time Clock, exactly as you have on your hardware. It *could* be a timer issue, except that i can't manage how to accelerate mi timer or to slow down my t1xxp driver... -- Stefano -- Outgoing mail is certified Virus Free. Checked by AVG Anti-Virus (http://www.grisoft.com). Version: 7.0.279 / Virus Database: 265.4.0 - Release Date: 18/11/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems using AGI-get_data - almost solved
Ok, it seems that by executing a Playback prior to GET DATA, you won't hear
the audio from get data a majority of the time. When I changed the playback
to stream_file, it worked. However, I don't hear the first please enter
your, I only hear card number, then press pound. Also, after I have
confirmed by the user that the PIN is correct, Asterisk plays Thank You and
then hangs up. It should execute a function to go validate the PIN, but it
doesn't.
I have enclosed my code below:
-- code --
#!/usr/bin/perl -w
use Asterisk::AGI;
use WWW::Curl::easy;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-setcallback(\mycallback);
# print STDERR AGI Environment Dump:\n;
#
# foreach $i (sort keys %input) {
#print STDERR -- $i = $input{$i}\n;
# }
my $userid = $input{'calleridname'};
my $exten = $input{'extension'};
open(fileOUT, /var/log/asterisk/calls.log);
$logtime = gmtime(time);
print fileOUT ---\n;
print fileOUT [$logtime]:Userid - $userid\n;
if($exten eq 'h') {
$exten = User hangup;
print fileOUT [$logtime]:Dialed Digits - $exten\n;
print fileOUT ---\n;
close(fileOUT);
exit;
}
else {
print fileOUT [$logtime]:Dialed Digits - $exten\n;
print fileOUT ---\n;
close(fileOUT);
}
if($exten eq '1000') {
$AGI-verbose(User wants IVR - So we give it to them!);
$AGI-verbose(Dialing to give IVR to enter a PIN!);
$AGI-set_callerid(1000);
$AGI-exec(Dial, Zap/g1/0032);
exit 1;
}
my ($sth, $query);
[DB removed]
$attempts = 0;
use Mysql;
$dbh = Mysql-connect($DBHost, $DBDatabase, $DBUser, $DBPassword);
HandleError( System, Fatal, DBConnectFail, *Data,
Unable to create connection to database: $error )
if( $error = Mysql-errmsg );
$query = select * from associations where userid = '$userid';
$sth = $dbh-query( $query );
if($sth-numrows 1) {
$AGI-verbose(Dialing to give IVR to enter a PIN!);
$AGI-set_callerid(1000);
$AGI-exec(Dial, Zap/g1/0032);
$AGI-hangup();
exit;
}
else {
$AGI-verbose(User $userid found!\n);
@fetched = $sth-FetchRow;
my($MAC, $PIN, $acc_code, $id, $datetime, $active) = @fetched;
if($exten == *99) { # User wants to change their PIN
get_pin($acc_code, $id, $MAC);
}
elsif($PIN == || $active == 0) {
get_pin($acc_code, $id, $MAC);
}
else {
$callerid = $acc_code.#$PIN;
$AGI-verbose(Callerid : $callerid);
$AGI-set_callerid($callerid);
#$AGI-set_callerid(1000);
$AGI-verbose(Dialing $exten\n);
$AGI-exec(Dial, Zap/g1/0032);
#$AGI-exec(Dial, Zap/g1/***01);
#$AGI-exec(Dial, Zap/g1/0032);
exit;
}
}
sub mycallback {
my ($returncode) = @_;
print STDERR CALLBACK: User Hangup ($returncode)\n;
exit($returncode);
}
sub get_pin($$$) {
$account_code = $_[0];
$userid = $_[1];
$MAC = $_[2];
$attempts++;
if($attempts eq 3) {
$AGI-exec(Playback, thank-you-for-calling);
sleep(1);
$AGI-exec(Playback, goodbye);
sleep(1);
$AGI-hangup();
exit 1;
}
$AGI-noop();
$AGI-stream_file(please-enter-your);
$AGI-noop();
$AGI-exec(Playback,card-number);
#$AGI-exec('Playback', 'card-number');
# $AGI-exec(Playback, then-press-pound);
# $AGI-exec(Read, PIN, then-press-pound, 13);
# $AGI-exec(SetVar, PIN, PIN);
# my $pin = $AGI-get_variable('PIN');
my $pin= $AGI-get_data(then-press-pound, 1, 13);
$AGI-say_digits($pin);
$AGI-exec('Playback', 'if-correct-press');
$AGI-exec('SayNumber','1');
$AGI-exec('Playback', 'otherwise-press');
$AGI-exec('SayNumber','2');
my $correct= $AGI-get_data(then-press-pound, 1, 2);
if($correct eq 1) {
$AGI-exec(Playback,auth-thankyou);
#$AGI-exec(Playback,pls-stay-on-line);
my $return_code = validate_pin($pin, $account_code, $userid, $MAC);
$AGI-verbose(Return code: $return_code\n);
if($return_code eq 100) {
# $query = update associations set PIN = '$pin' where userid =
'$userid' and MAC = '$MAC';
# $stah = $dbh-query($query);
$AGI-exec(Playback, pin-number-accepted);
$AGI-hangup();
}
}
else {
$attempts = $attempts - 1; # Don't count this attempt against them
get_pin($account_code, $userid, $MAC);
}
}
sub validate_pin() {
$pin = $_[0];
$account_code = $_[1];
$userid = $_[2];
$MAC = $_[3];
$url = [removed]
$postfields = [removed]
$rawdata = ;
post_data($url, $postfields);
if($rawdata =~ m/true/) {
$return_code = 100;
return $return_code;
}
else {
$AGI-exec(Playback, pin-number-invalid);
$AGI-exec(Playback, pls-try-again);
get_pin($account_code, $userid, $MAC);
}
}
sub post_data($$) {
$url = $_[0];
$postfields = $_[1];
print STDERR $url\n$postfields\n;
my $curl =
Re: [Asterisk-Users] Fedora Core 3 supported?
Assuming that they do not does anybody have a set for Fedora core 2? Unfortunately I don't have the Hardware to go with it just playing and testing the server and yes I'm using it on FC2. It compiled fine and was able to connect to the testing server useing CLI. -- Mike Ramirez [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 for *
On 19/11/2004 22:44 Steve Underwood said the following: as a gateway between what ? if it's SS7 on one side, what's on the other ? SIGTRAN (SS7 over IP) on top of SCTP ? Yep, that kind of gateway. He has his own SCTP, and doesn't use the native Linux 2.6 one. in which case, if * got itself a SIGTRAN channel (i'm speculating here), then it'd be possible for * -- SIGTRAN -- OpenSS7 -- SS7 -- Some Node to work then, would it ? (my SS7 kungfu is virtually non-existent !) Its GPL, but you need a password for CVS, and that costs $1k. Since its hehehe, interesting concept, i guess. :) I understand people have TE405P running on BSD now. If that is correct, there shouldn't be a lot else to do. steve, i'll send you private email regarding this. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 3 supported?
Fred Skrotzki wrote: Is Fedora Core 3 Supported? Fred, I've just installed FC3 on a new box and will be installing Asterisk today. I've done it a couple times and had no problems with the compile and install. Just starting to learn *. I haven't gone beyond the compile/install and play the demo. Doug ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 for *
Dinesh Nair wrote: On 19/11/2004 22:44 Steve Underwood said the following: as a gateway between what ? if it's SS7 on one side, what's on the other ? SIGTRAN (SS7 over IP) on top of SCTP ? Yep, that kind of gateway. He has his own SCTP, and doesn't use the native Linux 2.6 one. in which case, if * got itself a SIGTRAN channel (i'm speculating here), then it'd be possible for * -- SIGTRAN -- OpenSS7 -- SS7 -- Some Node to work then, would it ? Well, now Linux 2.6 has SCTP.. :-) Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AgentMonitorOutgoing = is there an opposite ?
We are running a call queue - with, say, 5 agents, and have a requirement to record all agents calls. Incoming calls to a queue (555-1234) are being monitored correctly outgoing calls from an agents extension (where they have logged on) using AgentMonitorOutgoing are being recorded correctly However, is there a function (AgentMonitorIncoming) to check to see if the extension being called is an agent extension, and start the agentmonitor, as the AgentMonitorOutgoing does ? There are some times where the extension is being called directly, and thus bypassing the incoming call queue. Or will I have to do this via a dial plan ? Thanks in advance. Julian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unpredictables Hangups
Eric, What about some consulting in Metairie. We are working with asterisk in our Metairie office and could use some consulting. Can you help us? Ron Original Message From: [EMAIL PROTECTED] To: [EMAIL PROTECTED], [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Unpredictables Hangups Date: Fri, 19 Nov 2004 08:25:20 -0600 Near New Orleans Louisiana, but I am interested in long term, part time consulting work in the Toronto, ON area. [EMAIL PROTECTED] wrote: Eric, What state are you in? Ron Original Message From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Unpredictables Hangups Date: Fri, 19 Nov 2004 08:10:32 -0600 Stefano Finetti wrote: From: Eric Wieling [EMAIL PROTECTED] Google: Results 1 - 10 of about 149 from lists.digium.com for Unknown error 500. Specifically: http://lists.digium.com/pipermail/asterisk-users/2004-April/042912. ht ml http://lists.digium.com/pipermail/asterisk-users/2003-November/0281 05 .html http://lists.digium.com/pipermail/asterisk-users/2003-November/0281 05 .html http://lists.digium.com/pipermail/asterisk-users/2004-April/043860. ht ml http://lists.digium.com/pipermail/asterisk-users/2003-July/016538.h tm l http://lists.digium.com/pipermail/asterisk-users/2004-June/049410.h tm l http://lists.digium.com/pipermail/asterisk-users/2004-July/055478.h tm l http://lists.digium.com/pipermail/asterisk-users/2004-March/040502. ht ml Usually I require dinner and drinks before this kind of heavy duty handholding. Today I guess I was just feeling sorry for someone that can't google, so I figured I'd just give you a freebie. Well Eric... The fact is that i googled a lot. Read ALL the post you just linked, and a lot more. I've tried almost all that solutions... And if you had followed other pages on the google result you should have seen a thread opened by myself about 1 year ago on the list... Actually, i am in this situation: E100P with NO shared IRQs: [EMAIL PROTECTED] /root]# cat /proc/interrupts CPU0 0: 27710902 XT-PIC timer 1: 3 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 39199525 XT-PIC eth0 10: 276808484 XT-PIC t1xxp 11:4083904 XT-PIC Cyclades-PC300 12: 0 XT-PIC PS/2 Mouse 14: 237800 XT-PIC ide0 15: 3 XT-PIC ide1 NMI: 0 LOC: 27711957 ERR: 0 No USB enabled on Motherboard, IRQ fixed via Motherboard BIOS in order to be absolutely sure that it won't be shared with anything (as the /proc/interrupts output shows well) I still go Read on 32 and Read on 37 fails with unknown error 500. Using Zttool, i see some Lost Interrupt, but even if i reduced a lot the number of irq losses, they still there. It seems that there is no way to completely stop irq losses, but the problem is that actually i can't do anything to stop irq losses, because i ran out of ideas on how to solve this problem, and this is the reason why i was asking again here, in the hope of someone who found the same problem recently and found an appropriate solution. Following one of the advices from this thread, i checked for busydetect and busycount. Now i'm monitoring if calls are still dropped randomly. But it seems that PRI error and Call drops are not so-strictly linked... Did you mention your extensive google search in your message? If so I didn't notice it. Do you have a /etc/sysconfig/harddisks ? If so uncomment USE_DMA=1, MULTIPLE_IO=16, EIDE_32BIT=3, LOOKAHEAD=1 and add -u1 to EXTRA_PARAMS. If you already have those enabled, try commenting out all of them except the EXTRA_PARAMS=-u1 If all else fails replace the motherboard with something different (different brand/chipset/etc). Are you using any RAID? If so disable it. Several people have reported problems with Promise RAID that were solved when they removed the Promise RAID card. --Eric ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] Call pickup
- Original Message -
From:
Walt
Reed
To: Leandro
Cc: Walt Reed ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, November 16, 2004 2:11
PM
Subject: Re: [Asterisk-Users] Call
pickup
On Tue, Nov 16, 2004 at 01:26:22PM +0100, Leandro said: From:
"Walt Reed" [EMAIL PROTECTED]
On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said: I
don't understand how to get call pickup to work with asterisk.
Have I to define *8 extension in the dialplan? to what?
Have I to include something, like for parked call? Has the
stable 1.0.2 version the pickup group feature? or I need to
patch it with bristuff? Search the wiki for call
pickup. It's all there. Unfortunately I have already read all
the readable on wiki without understanding the needed steps to get
call pickup to work. Can you please answer my questions?What
particular part do you not understand?The first search result hit
describes call pickup in general.The second describes how to create
pickup groups. You need to do this.The third shows where *8 is defined
and that you can change it tosomething else. *8 has been built-into
asterisk for a very long time. In1.0.2 you can change it to some other
code.That's it. Once you have defined your groups for all the
differentchannels you have (SIP, Zap, IAX, etc.), it just works. If you
haveproblems, you will need to give detailed information on how you
haveyour groups set in all the various channels involved, log examples,
etc.Make sure you look at the example configuration files that come
withasterisk.
I really hate to ask silly questions and thank you
for your time, but pickup group doesn't work yet. Maybe the pickup doesn't work
across Zap channels.
This is what I get when Zap/25 is ringing Zap/14
and Zap/7 try to pickup. I get "invalid extension" when I press *8#
- Starting simple switch on
'Zap/25-1' -- Executing Answer("Zap/25-1", "") in new
stack -- Executing Dial("Zap/25-1", "Zap/14") in new
stack -- Called 14 -- Zap/14-1 is
ringing -- Executing DigitTimeout("Zap/7-1", "3") in new
stack -- Set Digit Timeout to 3 --
Executing ResponseTimeout("Zap/7-1", "10") in new stack --
Set Response Timeout to 10 -- Zap/14-1 is
ringing -- Invalid extension '*' in context 'interno' on
Zap/7-1 == CDR updated on Zap/7-1 -- Executing
Playtones("Zap/7-1", "Congestion") in new stack -- Invalid
extension '8' in context 'interno' on Zap/7-1 == CDR updated on
Zap/7-1 -- Executing Playtones("Zap/7-1", "Congestion") in
new stack -- Invalid extension '#' in context 'interno' on
Zap/7-1 == CDR updated on Zap/7-1 -- Executing
Playtones("Zap/7-1", "Congestion") in new stack --
Zap/14-1 is ringing -- Hungup 'Zap/7-1'
This is my /etc/asterisk/zapata.conf
context=internosignalling=fxo_lsflash=100group=1callgroup=5pickupgroup=5channel
= 1-24
context=pstnsignalling=fxs_kscallgroup=5pickupgroup=5group=2channel
= 25
context=voipsignalling=fxs_kscallgroup=5pickupgroup=5group=3channel
= 26
This is the dialplan
[interno]include = parkedcalls
exten = t,1,Hangupexten =
i,1,Playtones(Congestion)
exten = s,1,DigitTimeout,3
exten = s,2,ResponseTimeout,10
exten =
4,1,Goto(componiinternoserie4,s,1)exten =
5,1,Goto(componiinternoserie5,s,1)exten =
6,1,Goto(componiinternoserie6,s,1)
exten = 0,1,Goto(impegnolinea,s,1)
exten =
3001,1,MusicOnHold()
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RE: [Asterisk-Users] Little off topic
Here is what I was trying to do Telco gives me a PRI It connects to my Asterisk via Wildcard T100P using a 2nd Wildcard T100P I would like to connect with a X-Cable my HylaFax server. My question is will a Wildcard T100P work in a Hylafax server? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, November 19, 2004 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Little off topic Martin List-Petersen wrote: Citat Eric Wieling [EMAIL PROTECTED]: Martin List-Petersen wrote: You can't, the T100P is a unchannelized T1 card. This is 100% wrong. The T100P supports Channelized Voice T-1 (aka CT1) If you want to use it with HylaFax you need either SpanDSP OR an analog port on Asterisk in addition to the T100P. Might be that i'm wrong on the unchannelized bit, but i don't see, where the analog port will help you ? The guy wants to do Hylafax directly on a T100P w/o Asterisk or Asterisk as middleware, which i don't see working. SpanDSP on the other side works well, but that is basically a softmodem emulation, something Hylafax can't do. I have not seen any applications for spandsp outside Asterisk, yet. *nod* I mist have missed the part about doing it all within Asterisk. I think I wrote that message before my 2nd cup of coffee. An analog port would allow you to plug a modem into the Asterisk box and run Hylafax using that. T-1- Asterisk - Analog - Modem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fedora Core 3 supported?
Sir, I am using FC3 with no problem. I have the T1 card. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Little off topic
Eric Hall wrote: Here is what I was trying to do Telco gives me a PRI It connects to my Asterisk via Wildcard T100P using a 2nd Wildcard T100P I would like to connect with a X-Cable my HylaFax server. My question is will a Wildcard T100P work in a Hylafax server? No it will not. Your only option is to use spandsp (see ftp.opencall,org) or an analog port in Asterisk with a modem and Hylafax. Spandsp's rx_fax and tx_fax just create .tiff files of the fax. You will have to write your own scripts to handle the files. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Little off topic
Your actual question then is can the zaptel driver be connected with to a faxgetty? faxgetty expects a serial port, if I am not mistaken. So, can zaptel give me a pseudo-serial port I can use with faxgetty? Not having tried it myself, my expectation would be that it can not. Greg Eric Hall wrote: Here is what I was trying to do Telco gives me a PRI It connects to my Asterisk via Wildcard T100P using a 2nd Wildcard T100P I would like to connect with a X-Cable my HylaFax server. My question is will a Wildcard T100P work in a Hylafax server? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, November 19, 2004 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Little off topic Martin List-Petersen wrote: Citat Eric Wieling [EMAIL PROTECTED]: Martin List-Petersen wrote: You can't, the T100P is a unchannelized T1 card. This is 100% wrong. The T100P supports Channelized Voice T-1 (aka CT1) If you want to use it with HylaFax you need either SpanDSP OR an analog port on Asterisk in addition to the T100P. Might be that i'm wrong on the unchannelized bit, but i don't see, where the analog port will help you ? The guy wants to do Hylafax directly on a T100P w/o Asterisk or Asterisk as middleware, which i don't see working. SpanDSP on the other side works well, but that is basically a softmodem emulation, something Hylafax can't do. I have not seen any applications for spandsp outside Asterisk, yet. *nod* I mist have missed the part about doing it all within Asterisk. I think I wrote that message before my 2nd cup of coffee. An analog port would allow you to plug a modem into the Asterisk box and run Hylafax using that. T-1- Asterisk - Analog - Modem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Little off topic
Eric Hall wrote: Telco gives me a PRI It connects to my Asterisk via Wildcard T100P using a 2nd Wildcard T100P I would like to connect with a X-Cable my HylaFax server. My question is will a Wildcard T100P work in a Hylafax server? That depends on your definition of work. If you mean will it cause the machine to function properly, sure. If you mean will Hylafax be able to use it to send and receive faxes without additional help, the answer is no, as others have already told you. Hylafax needs FAX modems to talk to. The T100P card (and all Digium cards) does not have FAX modems on it. There is some work being done to make Asterisk be able to emulate class 1 FAX modems for Hylafax to talk to, in which case you'd be able to put Hylafax _and_ Asterisk on that box and then send/receive FAXes over the T100P card. That is not available today, though. For Hylafax, if you want lots of channels and DID support, you will have to use one of the supported T-1 FAX cards: Patton, Digi, Eicon, etc. They are all very expensive, so be warned. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best line protocol for T1
A T1 PRI should be B8ZS, ESF. The protocol can be either 5ESS or NI2(not IN2). Either will work, primarily both ends need to be setup for the same protocol, but I would go with NI2 as that is a more 'universal' procotol(not switch specific like 5ESS). Lyle - Original Message - From: Jon Bebeau To: [EMAIL PROTECTED] Sent: Friday, November 19, 2004 8:22 AM Subject: [Asterisk-Users] "Best" line protocol for T1 Hello all, I'm provisioning a T1-PRI for a Digium T410P with my local TELCO.The TELCO has asked me to picka line protocol and has theoption of several RBS protocols, like 5ESS (Lucent), IN2 and others. The switch is a 5ESS, but the "normal" (according to the sales rep) protocol is IN2. I see from the doc on zaptel, that many protocols are supported. As I have a choice, is one of the PRI protocols "better" that an another? I suspect that some protocols support features over another but can't find much on the specifics. Recommendations and reasons please. Jon ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice
My BroadVoice account has been down for over a week with neither an explanation nor a service credit. Our problems may be a little different though because I don't remember what happened when I tried to dial out. I know that I do get a Request Timeout error while trying to register though. On Fri, 2004-11-19 at 08:39 -0600, Tim Jackson wrote: Anybody else having broadvoice problems? -- Executing SetAccount(SIP/101-d03b, LD) in new stack -- Executing Dial(SIP/101-d03b, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 408 Request Timeout back from 147.135.0.128 == No one is available to answer at this time Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim Mattison [EMAIL PROTECTED] Mattison Rosenthal Consulting Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel init script
I had problems with the init script not working ing FC2 also. I fixed it by editing the init script and changing 'insmod' to 'modprobe'. Don't know if that will fix your problem or not, but it's worth a try. -- Jim Dossey Computer Services -- Original message -- From: WipeOut [EMAIL PROTECTED] I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having an issue with the zaptel init script.. If I run.. #modprobe zaptel #modprobe wcfxo #modprobe wcfxs .. from a command line it load and appears to be working fine.. If I try and use the init script I get errors about ZT_CHANCONFIG and the modules don't see to laod up.. Anyone got any pointers? I am running Fedora Core 2 with all the updates and I have an X100P and a TDM400P with a single FXS module.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call pickup
Hi,
Have you configured features.conf file? the line which
enabled call pickup is commented and you have to un comment the line for call
pickup to work. Also you can define the numbering for call pickup
there
Thanks.
Yusuf
Alakavuk
Teknik Danman - Technical
Consultant
Grid Biliim
Teknolojileri A..
Kutepe Mahallesi Leylak
Sokak
Murat Merkezi A Blok Kat:2
Daire:9
34387 ili stanbul
Türkiye
Tel :
+90 (212) 336 92 55
Fax : +90
(212) 266 25 50
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
LeandroSent: 19 Kasm 2004 Cuma 17:52To: Walt Reed;
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Call pickup
- Original Message -
From:
Walt
Reed
To: Leandro
Cc: Walt Reed ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, November 16, 2004 2:11
PM
Subject: Re: [Asterisk-Users] Call
pickup
On Tue, Nov 16, 2004 at 01:26:22PM +0100, Leandro said: From:
"Walt Reed" [EMAIL PROTECTED]
On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said: I
don't understand how to get call pickup to work with asterisk.
Have I to define *8 extension in the dialplan? to what?
Have I to include something, like for parked call? Has the
stable 1.0.2 version the pickup group feature? or I need to
patch it with bristuff? Search the wiki for call
pickup. It's all there. Unfortunately I have already read all
the readable on wiki without understanding the needed steps to get
call pickup to work. Can you please answer my questions?What
particular part do you not understand?The first search result hit
describes call pickup in general.The second describes how to create
pickup groups. You need to do this.The third shows where *8 is defined
and that you can change it tosomething else. *8 has been built-into
asterisk for a very long time. In1.0.2 you can change it to some other
code.That's it. Once you have defined your groups for all the
differentchannels you have (SIP, Zap, IAX, etc.), it just works. If you
haveproblems, you will need to give detailed information on how you
haveyour groups set in all the various channels involved, log examples,
etc.Make sure you look at the example configuration files that come
withasterisk.
I really hate to ask silly questions and thank you
for your time, but pickup group doesn't work yet. Maybe the pickup doesn't work
across Zap channels.
This is what I get when Zap/25 is ringing Zap/14
and Zap/7 try to pickup. I get "invalid extension" when I press *8#
- Starting simple switch on
'Zap/25-1' -- Executing Answer("Zap/25-1", "") in new
stack -- Executing Dial("Zap/25-1", "Zap/14") in new
stack -- Called 14 -- Zap/14-1 is
ringing -- Executing DigitTimeout("Zap/7-1", "3") in new
stack -- Set Digit Timeout to 3 --
Executing ResponseTimeout("Zap/7-1", "10") in new stack --
Set Response Timeout to 10 -- Zap/14-1 is
ringing -- Invalid extension '*' in context 'interno' on
Zap/7-1 == CDR updated on Zap/7-1 -- Executing
Playtones("Zap/7-1", "Congestion") in new stack -- Invalid
extension '8' in context 'interno' on Zap/7-1 == CDR updated on
Zap/7-1 -- Executing Playtones("Zap/7-1", "Congestion") in
new stack -- Invalid extension '#' in context 'interno' on
Zap/7-1 == CDR updated on Zap/7-1 -- Executing
Playtones("Zap/7-1", "Congestion") in new stack --
Zap/14-1 is ringing -- Hungup 'Zap/7-1'
This is my /etc/asterisk/zapata.conf
context=internosignalling=fxo_lsflash=100group=1callgroup=5pickupgroup=5channel
= 1-24
context=pstnsignalling=fxs_kscallgroup=5pickupgroup=5group=2channel
= 25
context=voipsignalling=fxs_kscallgroup=5pickupgroup=5group=3channel
= 26
This is the dialplan
[interno]include = parkedcalls
exten = t,1,Hangupexten =
i,1,Playtones(Congestion)
exten = s,1,DigitTimeout,3
exten = s,2,ResponseTimeout,10
exten =
4,1,Goto(componiinternoserie4,s,1)exten =
5,1,Goto(componiinternoserie5,s,1)exten =
6,1,Goto(componiinternoserie6,s,1)
exten = 0,1,Goto(impegnolinea,s,1)
exten =
3001,1,MusicOnHold()
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Re: [Asterisk-Users] Little off topic
On 2004.11.19 07:47 Eric Hall wrote: My question is will a Wildcard T100P work in a Hylafax server? This question would be best fielded on the [EMAIL PROTECTED] mailing list, but the simple answer to your question is, no. The real answer to your question, though is this: PRI - T100P - Asterisk - T100P - T1 Fax Card* - HylaFAX So you still need two T100Ps. One to bring in the T1 PRI, and the other to send out to the fax card. As for the T1 Fax Card HylaFAX will take anything that has drivers that present themselves as standard tty devices. My favorite is the Digi/Patton DataFire 2977, however the Eicon Diva Server also works. You can also use other T1 cards that support CAPI drivers, but then you need to use capi4hylafax, and I don't know anything about that. Lee. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Little off topic
Kevin P. Fleming wrote: Eric Hall wrote: Telco gives me a PRI It connects to my Asterisk via Wildcard T100P using a 2nd Wildcard T100P I would like to connect with a X-Cable my HylaFax server. My question is will a Wildcard T100P work in a Hylafax server? That depends on your definition of work. If you mean will it cause the machine to function properly, sure. If you mean will Hylafax be able to use it to send and receive faxes without additional help, the answer is no, as others have already told you. Hylafax needs FAX modems to talk to. The T100P card (and all Digium cards) does not have FAX modems on it. There is some work being done to make Asterisk be able to emulate class 1 FAX modems for Hylafax to talk to, in which case you'd be able to put Hylafax _and_ Asterisk on that box and then send/receive FAXes over the T100P card. That is not available today, though. Probably around the end of the year. I sort of have HylaFAX half working with spandsp now. For Hylafax, if you want lots of channels and DID support, you will have to use one of the supported T-1 FAX cards: Patton, Digi, Eicon, etc. They are all very expensive, so be warned. Yeah, pricy. Wait a few weeks :-) Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm to iaxComm
On Thu, 18 Nov 2004 17:23:28 -0800, Adam Fineberg [EMAIL PROTECTED] wrote: Having some trouble with segfaults and sound quality all of a sudden (since I recompiled from the latest source) when 2 iaxComm clients connect. First off immediately after the server reports: -- Attempting native bridge of IAX2/[EMAIL PROTECTED]:4569/1 and IAX2/4589/5 The iaxclient library is in flux right now. The echo cancellation code is likely the cause, although I have heard of some problems resolved by disabling speex. I'm going to try to post new linux and windows binaries for iaxcomm this weekend that disable echo cancellation, and prefer iLBC. If you want to try it out, I've just posted a binary based upon 12NOV2004 CVS modified as above. It's not listed on the web page, here's a direct link (not guaranteed past this Sunday): http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-test.exe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel init script
-- Original message -- From: WipeOut [EMAIL PROTECTED] I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having an issue with the zaptel init script.. If I run.. #modprobe zaptel #modprobe wcfxo #modprobe wcfxs .. from a command line it load and appears to be working fine.. If I try and use the init script I get errors about ZT_CHANCONFIG and the modules don't see to laod up.. Anyone got any pointers? I am running Fedora Core 2 with all the updates and I have an X100P and a TDM400P with a single FXS module.. Later.. I belive I have seen on the list where wcfxs has been changed to wctdm this may be your problem? John Millican --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.788 / Virus Database: 533 - Release Date: 11/1/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup
- Original Message - From: Yusuf Alakavuk To: 'Asterisk Users Mailing List - Non-Commercial Discussion' ; 'Walt Reed' Sent: Friday, November 19, 2004 5:02 PM Subject: RE: [Asterisk-Users] Call pickup Hi, Have you configured features.conf file? the line which enabled call pickup is commented and you have to un comment the line for call pickup to work. Also you can define the numbering for call pickup there Are you referring to pickupexten=*8? Thank you for your try, but unfortunately, I have already uncommented it in features.conf:-( ;; Sample Parking configuration; [general]parkext = 700 ; What ext. to dial to parkparkpos = 701-720 ; What extensions to park calls oncontext = parkedcalls ; Which context parked calls are in;parkingtime = 45 ; Number of seconds a call can be parked for ; (default is 45 seconds);transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call;courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call;adsipark = yes ; if you want ADSI parking announcements pickupexten = *8 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SBC VoIP Tariff to ISP's
FYI SBC Makes VoIP Moves SBC has indicated in an FCC filing that it plans to file a federal tariff that will establish fees to be paid by ISPs that deliver VoIP calls to SBC's circuit switched end users. This service would not be mandatory. The rates for this service would be higher than the current reciprocal compensation rates paid for terminating local traffic, but lower than access charges applied to long distance calls. This tariff would be the first of its kind and could encounter opposition from the FCC, which is still in the process of finalizing rules relating to VoIP and intercarrier compensation issues. Separately, SBC has announced that it plans to launch residential VoIP services in all its markets in early 2005. These services are designed to build on SBC's base of DSL subscribers. SBC also announced that it has won what it claims is one of the nation's largest hosted VoIP contracts for the University of Notre Dame and has signed more than 450 contracts, valued at $1 million or more in the 3rd quarter. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Alcatel PBX
Dear Users, i have the following scnario. 1. Alcatel PBX with e1 module 2. asterisk server with 2 digium e100p cards. 1 connected to pstn e1, 1 connected to alcatel pbx. i m having problem in outgoing from alcatel. incoming from pstn - asterisk - alcatel working fine, but outgoing from alcatel - asterisk - pstn or any sip extensions not working. it hangs up the line as soon as i answer the call. i have generated dialtone via playtones but it has also issue. when i connect pstn e1 line directly to altacel e1 module, it works fine, but behind asterisk it hangups. any body have good idea ? further details can be provided if u need more. regards. -Neo This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Tecom IP2005 phone, problems :(
Hi, I'm having terrible trouble getting a Tecom IP2005 Sip phone working with Asterisk 1.0 I installed Asterisk couple weeks ago, then installed a X100P card and tested with X-Link softphone, all seemed well. So I thought I would buy a Sip phone from a UK company. However I cannot seem to get it to authorise with Asterisk. This is a link to the mfcr website :- http://www.tecomproduct.com/IP2005.htm And a link to the UK suppliers site:- http://www.solwise.co.uk/voip-phones-ip2005.htm Now with sip debug on I see messages like this: Sip read: REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.245:5060;branch=z9hG4bKcKaaZI5Ft Max-Forwards: 70 User-Agent: Centrality PA1688 From: home sip:[EMAIL PROTECTED];tag=yoyzIb5v2ZNzx08i To: home sip:[EMAIL PROTECTED] Call-ID: kv3Hc37gOQL6pI4k CSeq: 17455 REGISTER Contact: sip:[EMAIL PROTECTED]:5060 Expires: 360 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.1.245 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.245:5060;branch=z9hG4bKcKaaZI5Ft From: home sip:[EMAIL PROTECTED];tag=yoyzIb5v2ZNzx08i To: home sip:[EMAIL PROTECTED];tag=as249efa19 Call-ID: kv3Hc37gOQL6pI4k CSeq: 17455 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.1.245:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.245:5060;branch=z9hG4bKcKaaZI5Ft From: home sip:[EMAIL PROTECTED];tag=yoyzIb5v2ZNzx08i To: home sip:[EMAIL PROTECTED];tag=as249efa19 Call-ID: kv3Hc37gOQL6pI4k CSeq: 17455 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=4532aca5 Content-Length: 0 to 192.168.1.245:5060 Scheduling destruction of call 'kv3Hc37gOQL6pI4k' in 15000 ms splat*CLI The phone itself just displays Failed login message. The phone did come with some firmware which is supposed to give it SIP functionality, I've loaded this on and configured the sip server 192.168.1.2 in the phone. The phone IP is 192.168.1.245. Here is the section from sip.conf [home] type=friend username=home secret=secret callerid=home1 14 ;host=dynamic port=5060 defaultip=192.168.1.245 nat=no dtmfmode=rfc2833; Choices are inband, rfc2833, or info canreinvite=yes; Typically set to NO if behind NAT disallow=all allow=g723.1 allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw allow=alaw context=sip mailbox=2002 I'd appreciate any help! Many thanks, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel init script
[EMAIL PROTECTED] wrote: I had problems with the init script not working ing FC2 also. I fixed it by editing the init script and changing 'insmod' to 'modprobe'. Don't know if that will fix your problem or not, but it's worth a try. -- Jim Dossey Computer Services Hi Jim, Thanks for that, it has solved the problem.. -- Original message -- From: WipeOut [EMAIL PROTECTED] I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having an issue with the zaptel init script.. If I run.. #modprobe zaptel #modprobe wcfxo #modprobe wcfxs .. from a command line it load and appears to be working fine.. If I try and use the init script I get errors about ZT_CHANCONFIG and the modules don't see to laod up.. Anyone got any pointers? I am running Fedora Core 2 with all the updates and I have an X100P and a TDM400P with a single FXS module.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice
/SNIP/ My BroadVoice account has been down for over a week with neither an explanation nor a service credit. Our problems may be a little different though because I don't remember what happened when I tried to dial out. I know that I do get a Request Timeout error while trying to register though. Anybody else having broadvoice problems? /SNIP/ This is what happens to the VOIP Industry over time - Consolidation, if you want to call it that. A few players will remain at the end and when it is all over. All others will just disappear. Broadvoice has not understood the game it appears. This game is like Heavy Weight Boxing, where the last one standing is the winner. In order to be the last one to be standing, you have to be lean and mean and control your costs and keep acquiring customers, till you reach a critical mass and a value proposition for another investor or a predator. Anyone who has not understood these couple of things will fall by the wayside. Seshu Kanuri 732-213-2422 http://ipphone.eezeephone.com NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] i swtiched to digest
FuturaHost.Com Lists [EMAIL PROTECTED] wrote: I believe the list is so big that many of us are loosing some interesting threads. May be the admins can split the users list in some more specific sub-lists, and the people who wants to receive all the messages can subscribe to the sublists, or have a digest for someones, etc. You'll find that many people will want to be subscribed to all of the mail lists - just in case something interesting is said or asked. You'll also find that some Muppets will post their questions to multiple lists, instead of finding a specific list to use, or will judge that their question/comment has relevance in multiple lists. For instance, how many Asterisk veterans are likely to hang out on asterisk-newbies so that they can answer the same FAQ question every ten minutes, and how many Asterisk users are going to post their questions to the newbie list when they find that there are no experts there? People already post and/or duplicate end-user questions to the developers' list as it is. Splitting up the mail lists will most likely result in more bandwidth use for most people - not less. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel init script
John Millican wrote: -- Original message -- From: WipeOut [EMAIL PROTECTED] I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having an issue with the zaptel init script.. If I run.. #modprobe zaptel #modprobe wcfxo #modprobe wcfxs .. from a command line it load and appears to be working fine.. If I try and use the init script I get errors about ZT_CHANCONFIG and the modules don't see to laod up.. Anyone got any pointers? I am running Fedora Core 2 with all the updates and I have an X100P and a TDM400P with a single FXS module.. Later.. I belive I have seen on the list where wcfxs has been changed to wctdm this may be your problem? John Millican --- I will have to look into that, is appears to still be loading wcfxs according to the init script but maybe it hasn't been updated yet.. I guess thats the problem with being away from this for a while, things change so fast.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] i swtiched to digest
On November 19, 2004 11:43 am, Kevin Walsh wrote: You'll find that many people will want to be subscribed to all of the mail lists - just in case something interesting is said or asked. Personally I subscribe to -users, -dev and -cvs. You'll also find that some Muppets will post their questions to multiple lists, instead of finding a specific list to use, or will judge that their question/comment has relevance in multiple lists. For instance, how many Asterisk veterans are likely to hang out on asterisk-newbies so that they can answer the same FAQ question every ten minutes, and how many Asterisk users are going to post their questions to the newbie list when they find that there are no experts there? People already post and/or duplicate end-user questions to the developers' list as it is. This is exactly the problem with having a bazillion lists (as well as one of the problems with forums in general, IMO). Splitting up the mail lists will most likely result in more bandwidth use for most people - not less. Precisely. I see nothing wrong with what we have now (at least with the 3 I subscribe to) -- if you (the OP) are missing interesting threads, beat on the clueless ones who post with subject lines like I need help or asterisk problem -- I think the larger lists force the untrained to learn quickly or leave. The signup page should have the posting guidelines and a big fat warning that says FAILURE TO ADHERE TO THESE GUIDELINES WILL HAMPER YOUR ABILITY TO RECEIVE HELP -- Then let them whine on -newbies as to why they can't get anyone to help them. Too much hand-holding results in a mawkish society. People need to learn how to effectively communicate and stay afloat in the sea of information, not try and backfill the waters so they can feel comfortable. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 3 supported?
- Original Message - From: Jerry Geis [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 19, 2004 10:52 AM Subject: [Asterisk-Users] Fedora Core 3 supported? | Sir, | | I am using FC3 with no problem. I have the T1 card. | Has Core 3 been made to behave like Core 1 with respect to the zaptel drivers? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DS3 PCI in asterisk
On Nov 18, 2004, at 8:25 PM, Steven Critchfield wrote: Could someone get their hands on the driver to give it a good look and inform of licensing. IT mentions linux, and it mentions that it is channelized down to 672 DS0s. Sounds like the perfect card. Also, since you can get PCI-PMC carrier cards, this wouldn't require a cPCI system for testing. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] differents contexts for a channel
Hi all, I have a tdm04b card with 4 fxo's connected to 4 POTS of a media gateway. Supposing that I want to place the following calls: Zap/1 dials Zap/2 (by placing in /spool/outgoing a call file which dials a number corresponding to Zap/2) Zap/3 dials Zap/2 (also placing another call file) Zap/4 dials Zap/2 (also placing a call file) That's clear, in each situation Zap/2 will enter its incoming context defined in zapata.conf. My question is if it's possibile to change at runtime the context that Zap/2 uses ? If in Zap/2 to call comes from Zap/1 to enter a context and if the call comes from Zap/4 to enter another... Any help will be very much appreciated !! Thank you all for your time.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7970 Non-SIP Phone setup with Asterisk
Has anyone had any success setting up a 7970 to work with asterisk. I have searched all over and not found very much. Any advise would be greatly appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
